webrtc/modules/audio_coding/codecs
Karl Wiberg 918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
..
cng Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
g711 Extract third party part of g711 codec into separate target 2018-06-25 11:26:59 +00:00
g722 Extract third party part of g722 codec into separate target 2018-06-25 11:30:59 +00:00
ilbc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
isac Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
pcm16b Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
red Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
tools Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_decoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_format_conversion.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_format_conversion.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
builtin_audio_decoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
builtin_audio_encoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
legacy_encoded_audio_frame.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00